The present invention relates to radiotelephone communication systems and more particularly to a method and apparatus for reducing the delay in transmitting a block of speech to a mobile station, the delay being associated with the use of time division multiple access transmission of speech parameters on a pulse code modulation link between a mobile services switching center and a base station.
A cellular telephone communications system may include elements as shown in FIG. 1. Within the Public Land Mobile Network (PLMN) 100, a base station (BS) 103 is connected to a mobil services switching center (MSC) 101 by means of a first Pulse Code Modulation (PCM) link 107. The MSC is then linked to a Public Switched Telephone Network (PSTN) 115 by means of a second PCM link 113.
The MSC 101 has a Group Switch (GS) 111 for properly routing speech between the PSTN 115 and the appropriate base station 103. The MSC 101 also includes a Transcoder and Rate Adaptor (TRA) 109 which performs speech coding/decoding (data compression/expansion) and data rate adaptation of the speech signals that flow between the PSTN 115 and the base station 103. In operation, speech received by the MSC 101 via the second PCM link 113 is first compressed by the TRA 109, and then transmitted to the base station 103 by means of the first PCM link 107. This transmission preferably uses the error detection and handling mechanisms disclosed by commonly owned U.S. patent application Ser. No. 08/085,044, entitled "PCM LINK FAULT DETECTION AND LOST BLOCK HANDLING" by P. Sellin et al., filed on Jul. 2, 1993, the entire disclosure of which is hereby incorporated by reference. After it is received by the base station 103, the encoded speech is relayed to a mobile station (MS) 105. A speech coder-decoder (codec) (not shown) inside the mobile station 105 decodes the received speech parameters as part of the process of generating audible speech.
A similar set of operations takes place in the opposite direction. That is, speech originating in the mobile station 105 is encoded by the codec (not shown) located in the mobile station 105, and then transmitted to the base station 103. The encoded speech is then relayed, by means of the first PCM link 107, to the MSC 101. Inside the MSC 101, the encoded speech is decoded by the TRA 109, and then forwarded to the PSTN 115 by means of the GS 111 and the second PCM link 113.
The first PCM link 107 transmits data at a rate of 64 kilobits per second (kbit/s). However, this capacity is effectively increased by the fact that the speech being conveyed is compressed (i.e. encoded) as described above. The type of compression which is used in cellular telephone systems, such as the European Global System for Mobile Communication (GSM) or the American Digital Cellular (ADC) system, permits the 64 kbit/s connection to be shared by more than one voice and/or data channels (four channels for GSM, and three channels for ADC). It should be noted that this sharing is not possible on the second PCM link 113, which connects the PLMN 100 with the PSTN 115, because the signals conveyed on this link are not compressed. In practice, the MSC 101 is connected to multiple second PCM links 113, so that the MSC 101 may handle more than one connection at a time.
To understand the problem with the existing system, it is necessary to know how speech is encoded/decoded, how encoded speech/data from more than one connection is formatted for transmission on the first PCM link 107, and how encoded speech/data from more than one connection is formatted for transmission from the base station 103 to the mobile station 105. These topics will now be briefly discussed.
The TRA 109 collects 160 samples from each 20 ms of speech arriving from the PSTN 115 by means of the second PCM link 113. After the entire block of 160 speech samples has been collected, it is then processed and output from the TRA 109 as encoded speech. When an entire block of encoded speech has been collected, it is available for transmission to the base station 103 by means of the first PCM link 107. A block of encoded data consists of at least 159 bits of speech parameters.
As previously stated, the compression (encoding) of speech effectively increases the capacity of the first PCM link 107, so that it may be shared by more than one connection at a time. In known systems, such as GSM, this has been accomplished by formatting transmitted data as shown in FIG. 2. For every byte (8 bits) 201 of transmitted data, each of four users is allocated 2 bits. As illustrated in FIG. 2, User 1 transmits 2 bits of encoded speech and/or control data in bits B0 and B1, User 2 transmits 2 bits of encoded speech and/or control data in bits B2 and B3, User 3 transmits 2 bits of encoded speech and/or control data in bits B4 and B5, and User 4 transmits 2 bits of encoded speech and/or control data in bits B6 and B7. Then, the process starts all over again, with User 1 transmitting the next 2 bit of encoded data and/or control data in bits B0 and B1, and so forth.
At the base station 103, data from the first PCM link 107 is collected until a complete block of encoded speech and/or data has been accumulated. The encoded speech portion of a complete block is 20 bytes long in the American PCM system (D-AMPS), and 260 bits (33 whole bytes) in the European GSM system. Then, the complete block of encoded speech or data is processed by a channel codec 117 which adds data for error detection and correction, and then reformats the data into the format shown in FIG. 3. This format, shown in FIG. 3, is used for transmitting data over the "air interface," that is, from the base station 103 to the mobile station 105. As illustrated, the air interface of known mobile telephone communications systems is oriented around 20 ms frames 301, each of which can be shared by three users. In the example shown, User 1 transmits 324 bits of its data (including 312 bits representing information of some kind) during the first 6.67 ms 303 of the air interface frame 301, User 2 transmits its data during the second 6.67 ms 305 of the frame 301, and User 3 transmits its data during the third 6.67 ms 307 of the frame 301. Then the process is repeated for the next air interface frame 301. This method of sharing the use of a single resource (i.e., the air interface frame) is commonly referred to as Time Division Multiple Access (TDMA), and is well known in the art.
The problem with the aforementioned system will now be presented. As described above, each of the TRA 109 and the channel codec 117 needs a complete block of data before any processing (i.e., data compression/expansion or reformatting) can be performed. It follows, then, that a complete block of data must be received from the first PCM link 107 before any of it can be processed. This introduces a delay equal to the transfer time for a complete block. This transfer time delay may be expressed as: EQU delay=number of bits in block/channel capacity (1)
Equation (1) represents the delay that occurs when transmitting data in either direction between the PSTN 115 and the mobile station 105. In practice, a block of data consists of at least 159 bits of speech parameters, and more realistically may be 260 bits of user data plus approximately 32 bits of data for addressing and protection against errors on the PCM link. Also, for a connection with a nominal channel capacity of 64 kbit/s, the use of the channel formatting scheme shown in FIG. 2, in which a 64 kbit/s connection is shared by four multiplexed voice channels, results in an effective channel capacity equal to 1/4.times.64 kbit/s=16 kbit/s for each user. Consequently, the transfer delay in each direction may be computed to be between 159/16000=9.94 ms and 292/16000=18.25 ms.
Digital mobile telephony has quite long "built in" delays resulting from speech coding, channel coding, interleaving, and calculation times. The introduction of an extra delay for transfer of data from the TRA 109 to the base station 103 may lead to inconvenience when using the system, because the total delay exceeds what is acceptable.
Another drawback with the prior known system is the fact that if it is desired to multiplex a number of users that is not evenly divisible into 8 (since multiplexing is performed on a byte-by-byte basis), transmission capacity of the first PCM link 107 is wasted.